[Asterisk-Dev] Petition for IAX firmware

[Paul]

Steve Kann wrote:

>
> On Apr 5, 2005, at 10:49 PM, Paul wrote:
>
>> denon wrote:
>>
>>> Hi all,
>>>
>>> I've put together a quick petition, in hopes that we can possibly 
>>> persuade Sipura (or any other large-scale IP handset manufacturer) 
>>> to include firmware support for IAX. The IAXy has proven that an IAX 
>>> product is in demand, and very useful, and I think we'd all like to 
>>> see a handset manufacturer follow Digium's lead. I'm not 
>>> particularly endorsing Sipura, however I do know that they have 
>>> seriously considered support for IAX, and have decided to hold off 
>>> until "the demand is there". I'm hoping that with some numbers, we 
>>> can prove to them that the demand is already here, and that IAX is 
>>> already a viable technology.
>>>
>>> I'd like to encourage everyone to show your support -- hopefully 
>>> Sipura, and/or other manufacturers will see these hard names and 
>>> numbers, and realize it's time to move something into production.
>>>
>>> Petition:
>>> http://www.petitiononline.com/IAXPhone
>>>
>>> Thanks,
>>>
>>> -d
>>
>>
>> I like to see IAX support in any phone or ata. As for the benefits of 
>> IAX trunking: Suppose that you have 4 SPA-2000 devices at a branch 
>> office with IAX trunking back to headquarters? You won't be getting 
>> the best bandwidth usage unless you also run a * server at the branch 
>> office to aggregate those 8 extensions into one trunk. Otherwise you 
>> have 4 trunks to the * server at headquarters. With 8 IP phones it's 
>> even worse.
>>
>> So it looks like I need to have a linux box running at each location 
>> where I want to use IAX trunking. I'm sure I can get the cost of that 
>> down to something reasonable. I guess the next question is: It is 
>> much easier to do IAX trunking if all the ata's and IP phones use 
>> IAX2 instead of SIP? A good example would be 8 IAXy devices as 
>> compared to 4 2-port SIP ata's. Is the configuration going ot be 
>> easier? How about performance and stability?
>>
>> Rather than petition Sipura, I would prefer that we convince Digium 
>> to go further with the IAXy concept. Give us a base unit that accepts 
>> a mix of fxo or fxs modules. Offer it in 2, 4 and 8 port versions. If 
>> those 3 sizes will do trunking and offer the best free codecs, it 
>> will eliminate the need for a * server to do trunking at a lot of 
>> locations. Digium should do it first and let the others play catch-up 
>> when they finally see the light.
>>
>
> I have two points to make:
>
> 1) If I were a vendor, and there was documentation on the IAX2 
> protocol (not even an RFC, but at least some kind of semi-official 
> documentation), I'd be a lot more likely to implement it.
>
> 2) The idea of an IAX2 trunk-aggregator is interesting.  This is 
> probably something that doesn't need a whole x86 linux box -- taking 
> multiple IAX2 streams and putting them into trunks is trivial in terms 
> of computational requirements..  A small microcontroller or an ARM 
> chip is more than enough for this, and would be a neat idea of virtual 
> PBX deployments..
>
> -SteveK

There are a few linux-compatible boards being used right now for 
applications such as wireless ISP's. A trunk-aggregator should not do 
any transcoding. That would be handled by the * server at the other end 
of the trunk(if needed). The dial plan is also handled by the "master 
server". And so on and so on. That means maybe we can do it with a 
fanless pentium-compatible board that boots from a compact flash card. I 
have seen fanless products ranging from 133 to 800 mhz cpu speed.

But I still don't know the answer to my question about SIP vs. IAX2. 
Suppose the remote site has a mix of SIP and IAX2 devices. Does the 
presence of SIP devices increase the computational requirements much? My 
thinking is that any SIP ata's or phones at the remote site are going to 
be extensions of the master * pbx. Hopefully that makes it easier on the 
trunk-aggregator cpu. Also I expect that in most situations where this 
was deployed a codec other than g.711 would be used since there is a 
motivation to conserve bandwidth. So we have SIP traffic from a provider 
to the master server, IAX2 trunking to the remote slave server and back 
to SIP over the LAN to somebody's deskset. Will that conversion back to 
SIP to reach the deskset degrade the call quality?